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| 00:18.37 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.21.0 (2018/05/01), Standard: 15.4.0 (2018/05/01); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 15:19.05 | kodomo | Hi all! Is it possible that chan_sip accepts INVITES without explicitly stated audio codecs, whereas pjsip won't? |
| 15:20.15 | Samot | No. |
| 15:20.25 | Samot | All INVITES should have SDP in them. |
| 15:21.12 | kodomo | I'm trying to place a call to a ConfBridge with a client (CSipSimple) which has been configured to use GSM (both the lient and asterisk in sip.conf and pjsip.conf) ; in chan_sip, the call goes through, in pjsip, it stops with an 488 Unnacceptable Here |
| 15:21.27 | kodomo | sdp I have - just not 'GSM' in the debug output |
| 15:21.54 | kodomo | or - sec... double checking |
| 15:24.27 | Samot | 488 means you have a codec mismach most likely. |
| 15:27.25 | kodomo | Here's a debug trace: https://pastebin.com/2856ejGZ |
| 15:27.41 | kodomo | Output goes straight from [May 27 16:20:59] DEBUG[25358]: res_pjsip_session.c:269 handle_incoming_sdp: Negotiating incoming SDP media stream 'audio' using audio SDP handler |
| 15:27.59 | kodomo | to : [May 27 16:20:59] DEBUG[25358]: res_pjsip_session.c:269 handle_incoming_sdp: Negotiating incoming SDP media stream 'audio' using audio SDP handler |
| 15:28.02 | kodomo | [May 27 16:20:59] DEBUG[25358]: res_pjsip_session.c:2517 handle_outgoing_response: Method is INVITE, Response is 488 Not Acceptable Here |
| 15:28.15 | kodomo | I don't see an additional error. |
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| 15:28.56 | kodomo | output from pjsip show endpoint 1244 contains allow : (gsm) |
| 15:28.59 | kodomo | <PROTECTED> |
| 15:29.09 | kodomo | So codec should be fine |
| 15:29.26 | Samot | https://www.irccloud.com/pastebin/ynT5OFYH/ |
| 15:30.38 | Samot | This debug is wrong. |
| 15:30.49 | Samot | It's not complete, you've cut out things. |
| 15:30.54 | Samot | asterisk -rvvvvvvvvvvv |
| 15:30.58 | Samot | pjsip set logger on |
| 15:31.05 | Samot | ^^ Run those and do it again |
| 15:31.24 | Samot | No debug |
| 15:31.34 | Samot | core set debug off |
| 15:32.23 | kodomo | It's a snippet from the trace, do you lack details or just the context around these lines? |
| 15:32.44 | Samot | That debug is not the full call. |
| 15:33.00 | Samot | You don't have the initial INVITE and you don't have the full 488 packet reply |
| 15:33.24 | kodomo | ok - sec... pasting the full trace... |
| 15:38.29 | kodomo | Samot: https://pastebin.com/VTJxZVCR |
| 15:39.51 | Samot | So either your crypto or codecs are wrong. |
| 15:39.52 | Samot | Or both |
| 15:40.19 | kodomo | They work with chan_sip (exactly the same config) |
| 15:40.28 | Samot | With TLS? |
| 15:40.32 | kodomo | with TLS |
| 15:40.34 | Samot | And it can't be exactly the same config. |
| 15:40.42 | Samot | Settings are not the same |
| 15:41.06 | kodomo | to the best of my knowledge - meaning: crypto keys are the same files, config contains tls and gsm only |
| 15:41.37 | Samot | Show the PJSIP config for the peer |
| 15:41.38 | file | TLS and SRTP are independent. |
| 15:41.45 | Samot | ^^^ |
| 15:42.20 | Samot | Which means the crypto or the codecs are wrong. |
| 15:42.33 | Samot | They don't match between what the client and what PJSIP is setup to deal with. |
| 15:47.40 | kodomo | Samot: https://pastebin.com/cGCJJet3 |
| 15:47.56 | kodomo | (It's slow, as I'm on a train => decent lag ;) ) |
| 15:48.33 | Samot | There is zero media encryption set for this peer. |
| 15:48.37 | file | Your endpoint does not have SRTP enabled, so the offer gets rejected |
| 15:48.38 | Samot | So it's not going to work. |
| 15:50.04 | kodomo | ah - so the encryption specified is only for signallint in pjsip? Then I need to figure out how to specify media encryption :) |
| 15:51.29 | kodomo | does merely setting media_encryption=yes suffice? (indeed, I just discovered that it was 'false' by default :) |
| 15:51.46 | file | No, you need to specify sdes |
| 15:51.51 | file | There are multiple types. |
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| 16:11.50 | Vooray | Which version should i start learning asterisk with? chan_sip or pjsip? Im new to all this voice things. |
| 16:16.43 | sibiria | pjsip. chan_sip won't be getting any new features |
| 16:18.20 | Vooray | Which asterisk version is used for new production deployments? |
| 16:18.48 | Vooray | they have too many... 13 certified, 14, 15 |
| 16:19.13 | sibiria | 13 is LTS. there's some benefit to that |
| 16:20.19 | Vooray | Thanks, sibiria |
| 16:21.09 | sibiria | get the non-certified 13, though |
| 16:22.25 | kodomo | Samot, file: thx for the help! I can see ConfBridge starting now :), !na3pt6 |
| 16:32.39 | *** join/#asterisk david71 (2eb9f0cd@gateway/web/freenode/ip.46.185.240.205) |
| 16:33.50 | david71 | Dears Hope you all well , After spending 1 week on google i decided to ask the experts here ,, im using the freepbx asterisk to make calls from chan_mobile using the bluetooth , I have connected the iphone and the calls is passing normally i was wondering why i can send letter in asterisk |
| 16:34.32 | david71 | im trying to send call facetime://number here so the call will go through facetime using my bluetooth but asterisk changes the words to numbers and send the call anyway to send the call and run facetime |
| 16:43.18 | Samot | FaceTime is incompatible with no FaceTime devices. |
| 16:56.12 | david71 | Samot but we are sending the calls from bluetooth |
| 16:56.35 | Samot | facetime:// == Apple Only |
| 16:56.40 | david71 | if i do it manual on iphone i will have the option to choose if i want call on facetime or regular call , but when we send from asterisk it doesnt work |
| 16:56.47 | Samot | There is no standard facetime:// protocol to use for other devices. |
| 16:57.18 | david71 | man if i make manual call on facetime and connect bluetooth it will work |
| 16:57.20 | Samot | facetime:// is an iOS/OSX thing. |
| 16:57.29 | david71 | but when the call is dialed from asterisk no option to choose facetime |
| 16:57.39 | Samot | Sigh |
| 16:57.45 | Samot | Asterisk does not support it. |
| 16:57.55 | Samot | Asterisk cannot talk/communicate over facetime:// |
| 16:58.08 | david71 | so if i want this to happen i will have to make iphone choose facetime for default calls |
| 16:58.26 | david71 | i have done it on some saved contacts numbers |
| 16:58.33 | david71 | because you can set it as default |
| 16:58.38 | david71 | to choose facetime |
| 16:58.39 | Samot | So? |
| 16:58.42 | david71 | but i want it for all calls |
| 16:58.52 | Samot | You're talking about a setting on the iPhone. |
| 16:59.00 | Samot | Asterisk _does not_ support FaceTime |
| 16:59.01 | david71 | not for the contact i edit default calls |
| 16:59.10 | Samot | Non-Apple devices do not support FaceTime |
| 16:59.19 | Samot | They cannot accept or make calls over that protocol. |
| 16:59.35 | david71 | there must be a way through bluetoooth |
| 16:59.58 | Samot | Have you found one? |
| 17:00.12 | david71 | not really |
| 17:00.12 | Samot | You said you spent a week on Google and now you're here to ask the experts. |
| 17:00.16 | file | chan_mobile is a community supported rarely used module, it is unlikely you will find anyone with knowledge or information about such a thing |
| 17:00.24 | Samot | You are now being told it's not supported. |
| 17:01.45 | Samot | FaceTime uses ports 53, 80, 443, 4080, 5223, and 16393-16472 (UDP) <-- Nothing any real SIP system supports for SIP |
| 17:02.11 | Samot | Outside of port 53 for DNS lookups when needed. |
| 17:03.41 | file | Samot: you don't speak the "protocol" over Bluetooth, you use some AT based commands to tell the phone to do things and then it does them |
| 17:03.52 | Samot | OK. |
| 17:04.09 | file | somehow doubts the AT commands would do Facetime dialing |
| 17:04.47 | Samot | I don't think it would. |
| 17:04.52 | Samot | FaceTime is not open source |
| 17:05.10 | file | it's also an application running on the phone, not part of the phone stack itself |
| 17:05.20 | Samot | True. |
| 17:05.54 | file | the use of the "facetime" URI is likely just a hint to the interface in the phone to use the facetime application for the specific URI, and then it gets passed the data |
| 17:06.41 | Samot | Well I'm sure Asterisk is puking on the request if it's turning letters into numbers. |
| 17:06.58 | Samot | Or something because sip:bob@ip:5060 works |
| 17:07.29 | file | the person who wrote chan_mobile may have put in code to have it convert it into numbers |
| 17:07.53 | Samot | Possible. I've never used it |
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